added some more ASIO sample conversions

src/client.h

@@ -142,7 +142,7 @@ public:
         // sound card conversion buffer is used or not
         if ( bSndCrdConversionBufferRequired )
         {
-            return iSndCardMonoBlockSizeSamConvBuff;
+            return iSndCardMonoBlockSizeSamConvBuff + iMonoBlockSizeSam;
         }
         else
         {

src/clientsettingsdlg.cpp

@@ -8,16 +8,16 @@
  *
  * This program is free software; you can redistribute it and/or modify it under
  * the terms of the GNU General Public License as published by the Free Software
- * Foundation; either version 2 of the License, or (at your option) any later 
+ * Foundation; either version 2 of the License, or (at your option) any later
  * version.
  *
- * This program is distributed in the hope that it will be useful, but WITHOUT 
+ * This program is distributed in the hope that it will be useful, but WITHOUT
  * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
- * FOR A PARTICULAR PURPOSE. See the GNU General Public License for more 
+ * FOR A PARTICULAR PURPOSE. See the GNU General Public License for more
  * details.
  *
  * You should have received a copy of the GNU General Public License along with
- * this program; if not, write to the Free Software Foundation, Inc., 
+ * this program; if not, write to the Free Software Foundation, Inc.,
  * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
  *
 \******************************************************************************/
@@ -103,12 +103,15 @@ CClientSettingsDlg::CClientSettingsDlg ( CClient* pNCliP, QWidget* parent,
         "<li>512 samples: This setting should only be used if only a very slow "
         "computer or a slow internet connection is available.</li>"
         "</ul>"
-        "Some sound card driver to not allow the buffer delay to be changed "
+        "Some sound card driver do not allow the buffer delay to be changed "
         "from within the llcon software. In this case the buffer delay setting "
         "is disabled. To change the actual buffer delay, this "
         "setting has to be changed in the sound card driver. On Windows, press "
         "the ASIO Setup button to open the driver settings panel. On Linux, "
         "use the Jack configuration tool to change the buffer size.<br>"
+        "If no buffer size is selected and all settings are disabled, a "
+        "unsupported bufffer size is used by the driver. The llcon software "
+        "will still work with this setting but with restricted performannce.<br>"
         "The actual buffer delay has influence on the connection status, the "
         "current upload rate and the overall delay. The lower the buffer size, "
         "the higher the probability of red light in the status indicator (drop "
@@ -159,7 +162,7 @@ CClientSettingsDlg::CClientSettingsDlg ( CClient* pNCliP, QWidget* parent,
     cbUseHighQualityAudio->setAccessibleName ( tr ( "Use high quality audio "
         "check box" ) );
 
-    // current connection status parameter 
+    // current connection status parameter
     QString strConnStats = tr ( "<b>Current Connection Status "
         "Parameter:</b> The ping time is the time required for the audio "
         "stream to travel from the client to the server and backwards. This "

windows/sound.cpp

@@ -639,7 +639,16 @@ void CSound::bufferSwitch ( long index, ASIOBool processNow )
 
                 case ASIOSTInt24MSB:
 // NOT YET TESTED
-	                // TODO
+                    for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
+                    {
+                        // because the bits are flipped, we do not have to perform the
+                        // shift by 8 bits
+                        int iCurSam = 0;
+                        memcpy ( &iCurSam, ( (char*) bufferInfos[i].buffers[index] ) + iCurSample * 3, 3 );
+
+                        vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
+                            Flip16Bits ( static_cast<int16_t> ( iCurSam ) );
+                    }
 	                break;
 
                 case ASIOSTInt32MSB:
@@ -655,32 +664,64 @@ void CSound::bufferSwitch ( long index, ASIOBool processNow )
 
                 case ASIOSTFloat32MSB: // IEEE 754 32 bit float, as found on Intel x86 architecture
 // NOT YET TESTED
-	                // TODO
+                    for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
+                    {
+                        vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
+                            static_cast<int16_t> ( static_cast<float> (
+                            Flip32Bits ( static_cast<int32_t*> (
+                            bufferInfos[i].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
+                    }
 	                break;
 
                 case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture
 // NOT YET TESTED
-	                // TODO
+                    for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
+                    {
+                        vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
+                            static_cast<int16_t> ( static_cast<double> (
+                            Flip64Bits ( static_cast<int64_t*> (
+                            bufferInfos[i].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
+                    }
 	                break;
 
                 case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment
 // NOT YET TESTED
-	                // TODO
+                    for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
+                    {
+                        vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
+                            static_cast<int16_t> ( Flip32Bits ( static_cast<int32_t*> (
+                            bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFF );
+                    }
 	                break;
 
                 case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment
 // NOT YET TESTED
-	                // TODO
+                    for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
+                    {
+                        vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
+                            static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
+                            bufferInfos[i].buffers[index] )[iCurSample] ) & 0x3FFFF ) >> 2 );
+                    }
 	                break;
 
                 case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment
 // NOT YET TESTED
-	                // TODO
+                    for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
+                    {
+                        vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
+                            static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
+                            bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFFF ) >> 4 );
+                    }
 	                break;
 
                 case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment
 // NOT YET TESTED
-	                // TODO
+                    for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
+                    {
+                        vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
+                            static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
+                            bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFFFF ) >> 8 );
+                    }
 	                break;
                 }
             }
@@ -823,7 +864,15 @@ void CSound::bufferSwitch ( long index, ASIOBool processNow )
 
                 case ASIOSTInt24MSB:
 // NOT YET TESTED
-	                // TODO
+                    for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
+                    {
+                        // because the bits are flipped, we do not have to perform the
+                        // shift by 8 bits
+                        int32_t iCurSam = static_cast<int32_t> ( Flip16Bits (
+                            vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] ) );
+
+                        memcpy ( ( (char*) bufferInfos[i].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
+                    }
 	                break;
 
                 case ASIOSTInt32MSB:
@@ -841,32 +890,80 @@ void CSound::bufferSwitch ( long index, ASIOBool processNow )
 
                 case ASIOSTFloat32MSB: // IEEE 754 32 bit float, as found on Intel x86 architecture
 // NOT YET TESTED
-	                // TODO
+                    for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
+                    {
+                        const float fCurSam = static_cast<float> (
+                            vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
+
+                        static_cast<float*> ( bufferInfos[i].buffers[index] )[iCurSample] =
+                            static_cast<float> ( Flip32Bits ( static_cast<int32_t> (
+                            fCurSam / _MAXSHORT ) ) );
+                    }
 	                break;
 
                 case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture
 // NOT YET TESTED
-	                // TODO
+                    for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
+                    {
+                        const double fCurSam = static_cast<double> (
+                            vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
+
+                        static_cast<float*> ( bufferInfos[i].buffers[index] )[iCurSample] =
+                            static_cast<double> ( Flip64Bits ( static_cast<int64_t> (
+                            fCurSam / _MAXSHORT ) ) );
+                    }
 	                break;
 
                 case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment
 // NOT YET TESTED
-	                // TODO
+                    for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
+                    {
+                        // convert to 32 bit
+                        const int32_t iCurSam = static_cast<int32_t> (
+                            vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
+
+                        static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
+                            Flip32Bits ( iCurSam );
+                    }
 	                break;
 
                 case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment
 // NOT YET TESTED
-	                // TODO
+                    for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
+                    {
+                        // convert to 32 bit
+                        const int32_t iCurSam = static_cast<int32_t> (
+                            vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
+
+                        static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
+                            Flip32Bits ( iCurSam << 2 );
+                    }
 	                break;
 
                 case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment
 // NOT YET TESTED
-	                // TODO
+                    for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
+                    {
+                        // convert to 32 bit
+                        const int32_t iCurSam = static_cast<int32_t> (
+                            vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
+
+                        static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
+                            Flip32Bits ( iCurSam << 4 );
+                    }
 	                break;
 
                 case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment
 // NOT YET TESTED
-	                // TODO
+                    for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
+                    {
+                        // convert to 32 bit
+                        const int32_t iCurSam = static_cast<int32_t> (
+                            vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
+
+                        static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
+                            Flip32Bits ( iCurSam << 8 );
+                    }
 	                break;
                 }
             }